Protools 192k

[GW]

"The Apogee Guide to Digital Audio" is a folder that explain this problematic in plain understandble terms.

I think 24/192 sounds like a great idea and storage and DSP won´t be a problem for long - history has shown us that... [img]images/icons/wink.gif[/img]

[emilano]

I'll look at that, thanks. But I'm always a bit skeptical when the people who wrote the thing are the same ones trying to sell you the thing.

[Jules]

I am told one area where one CAN hear the difference is in the highest frequencies generated by the string section of an orchestra.. this high frequency 'excitement' caused in the room by all the bows scrapeing across the strings all belending into one sound, is - complicated to say the least and hard to pick up well.
Certainly this is ONE area of the recording world where 192k might take off.
We all may get the benefit of it soon on movie soundtracks..

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[Shaun.O]

yeah i agree.

i thought that at 96k, having twice as many samples taken of the audio signal amounted to half the distance (in time) between those samples.

After the distortion introduced from quantisation occurs, a more acurate reproduction of the analog waveform results
since the angular lines ( or squaring off ) introduced by quantisation are less imposing because the extra samples provide 'steping stones' to round off the part of the waveform originally messed between the original lower sample (44.1k) rate points.

the other half of this issue is quantisation. With 24 bit the resolution (number of quantisation values) is high so the extra samples provided by 96k have lots of options of where to go, whereas back in the 16 bit days with far fewer quantisation values available, a 96k sample rate would be less significant due to a lot of the samples being quantised to the same value.

SO

[Park Seward]

It is a common misconception to think that more samples will more accurately describe a waveform. That is not so. Once a waveform is accurately described, more samples will not describe it more accurately.

A higher sampling rate lets you describe higher frequencies but does not make the lower frequencies have more detail or resolution.

Same with more bits. You have more signal to noise ratio but not more detail.

See "Topic: setting record levels into ProTools with the intent of mixing in ProTools" for all the info.

From Nika: "The one thing that gets lost in a lot of this is that bits does not equal "resolution". Bits equals "dynamic range". What this means is that you get no better signal quality when recording hot vs. recording with headroom, so long as the noisefloor of your music is above the noisefloor of your converters. You really just want to make sure that the entire dynamic range of what you want to record is being captured. If you're recording Nine Inch Nails, where there is a total dynamic range of 12dB, it doesn't matter where on the meters it registers, so long as it is above around -80db in 16 bit, or -110dB or so in 24bit. "

First, as illogical as it seems, bits does NOT equal better resolution of anything but the noise in its signal path. More bits ONLY equals dynamic range.

First we have to talk about the difference between "quantization steps" and "bits". The number of "quantization steps" from top to bottom is defined by how many bits we have. A 1 bit signal has two quantization steps (0 and 1) a 2 bit signal has four, a sixteen bit converter has 65,536, and a 24 bit converter has ~16,000,000. Remember that half of these steps are below the zero axis and the other half are above, so to look a 24 bit signal, the audio goes from -8,000,000 to +8,000,000. Cool so far?

Let's also define "noise" really quickly. Noise=white noise. Any other form of noise is considered noise WITH SIGNAL, and you'll get different results if you treat the "apparent" noise floor as the "actual" noise floor. The actual noise floor is where the signal actually drops off into white noise only. If your signal drops off into pink noise, or other filtered noise then you have not actually hit the noise floor yet.

"Signal to noise" ratio deals specifically with white, pure, natural noise. When discussions over signal to noise ratios are brought up it is implied that the noise being spoken of is the fundamental level of white noise.

Let's talk about a sine wave with a signal to noise ratio of 42db. This will take 7 bits, or 128 quantization points to accurately capture and reproduce this sine wave, accepting that a each bit gives us 6dB of dynamic range capabilities (a whole other lecture, but a commonly accepted point. Run with me...). This means that the signal will take up all of -64 to +64 quantization steps. Now, to hopefully answer your next question, the signal, when turned up to maximum in an 8 bit converter, will indeed register from -128 to +127, thus implying more "resolution". When put into a 16 bit converter, will indeed register from ~-32,000 to ~+32,000. This is an example where more "quantization points" are used to capture this audio. Unfortunately, though, it is specifically NOT more resolution. Let me try to explain why:

Even though we have essentially 65,000 points now to describe that sinewave, it is really divided up into 128 chunks of about 512 quantization steps. This is because we know that it is a 42dB signal, and a 42 dB signal can only be divided into 128 quantization steps by definition. It'd be like trying to draw a line on graph paper with a can of spray paint. The width of the spray is only so resolved. Attempting to resolve it further has you defining more than the spray actually yielded. Making more refined graph paper isn't going to help describe that artwork you did any better. The width of the spray here represents noise, and the resolution of the graph paper represents the number of quantization steps. So back to our situation, a 42dB signal can ONLY be divided into 128 steps, period. Trying to do more than that has you defining more accuracy than the signal has.

The signal is going to fall within that 512 quantization step window, but exactly where within that window is not important because the resolution within that window of 512 quantization steps only helps to describe the white noise. But since this is white noise, it is unnecessary to describe it with precision, as any random area within that window is really noise. What this means is that the behavior of the signal within that window of 512 quantization steps is totally random. *

So the signal will pass through all of these 128 groups of 512 quantization steps, but the fact that it does just that, and at what times it passes through these ranges is all we need to know. Exactly where within that window it passes is totally irrelevant and does not give us any better resolution of the actual signal itself.

Now if we turn the signal down 6db so that we are only using 15 bits of our 16bit system so that, even though the system is CAPABLE of 64,000 increments from top to bottom, we're now only using 32,000 of them to describe this sine wave. We still only have 128 quantization steps for the signal itself, each of which is now divided into 256 quantization steps for the noise.

But that 42db signal is only ITSELF ever quantized into 128 discreet increments, and thus its resolution does not change no matter how many quantization points we add. As I said before, all that does is give you better resolution of a totally random noise signal.

So again, we need to be clear about using the word "resolution". The audio signal itself never has any better resolution than the minimum amount necessary to describe it, which can be ascertained by it's dynamic range (or signal to noise ratio). Increasing the bit depth does, in no way, benefit the "resolution" of the signal, and thus we try to avoid using that word to describe the effect of adding bit depth. All that adding bit depth does is allow us to accurately record material with wider dynamic range.

Again, I don't care HOW we use the word "resolution". The SIGNAL itself does not have any additional "resolution", "quantization steps", "discreet measurement increments", or any other term to describe units of measurement when we increase the bit depth beyond what is necessary to accurately describe the signal.

This all explains that bits only tells you how much total dynamic range the system is capable of. Any use of the higher number of quantization points to try to increase the audio's "resolution" is futile as it only hopes to describe with accuracy the system's random noise. This all feeds back to my point that you only need to record as hot as the dynamic range of your music allows. Any more than that is unnecessary. Thus, as I often say, "turn it down, it'll be fine!"

Thanks to Nika for his explanation.

[Jules]

While you digital brains are on line..

What sort of tragedy is the SRC of 96k sessions down to 16 bit 44.1?

My Masterlink can do it, I suppose there will be a conversion program for the Mac to do it.

What is the difference between the way high end mastering houses do it and how it might get done in a pro/project studio?

TIA

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[SmlTwnGuy]

<BLOCKQUOTE><font size="1" face="Verdana, Arial">quote:<HR>Originally posted by Park Seward:
[QB

Same with more bits. You have more signal to noise ratio but not more detail.

From Nika: "The one thing that gets lost in a lot of this is that bits does not equal "resolution". Bits equals "dynamic range". What this means is that you get no better signal quality when recording hot vs. recording with headroom, so long as the noisefloor of your music is above the noisefloor of your converters. You really just want to make sure that the... [/QB]<HR></BLOCKQUOTE>


I wouldn't be quite so confident quoting Nika. Bob Katz, yes.

[JasonWorrel]

Noiz2,

Well, I think your argument was that at lower sampling rates, the sampled waveform gets jaggier (lower resolution, and less accurate). However, upon playback, those jaggies, which are really super high frequencies, are filtered out with a brick wall filter part of all DA converters. This filter, if designed well, should leave the original waveform accurately reproduced.

[Kenny Gioia]

If this is all true, then why don't they make MP3's at 44.1 and 4 bit that sound like CD quality.

And why did the SP-12 and SP1200 sound so bad. 8 bit samplers.

Not doubting I'm just ignorant.

[emilano]

lwilliam.

This is in response to your comment about all of the higher than 20k freuquencies interacting in a room of musicians to form "beat frequencies" that color the sound in a good way. From what I have read, that is true, this does happen. When you record at 44.1 or 48khz with a room mic you already do capture these beat frequencies, because, since we can hear them, they must be within the recording limits of 44.1k recording. The only way that recording at higher sampling rates helps out is when you are recording an ensemble with close mics, or doing overdubs. Because, theoretically, with a higher sampling rate, these beat frequencies can be created DURING MIXING...in other words, you then wouldn't need to capture everyone playing at the same time in the same room. Currently you can only capture them when you are recording with room mics that capture a whole band or ensemble, because with overdubs or close micing, the opportunity to have the higher freqs interact is lost when you filter out those high frequencies during recording through your low sample rate convertors. But with high sampling rates, you could theoretically allow them to interact DURING MIXING. (so long as your microphones and everything else in your signal chain allowed for those high frequencies to be passed. Your monitors however would not need to be able to reproduce the high freqs).

Sounds good in theory anyway.